/*
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#include"config.h"
#ifENABLE(WEB_AUDIO)
#include"ReverbConvolver.h"
#include"VectorMath.h"
#include"AudioBus.h"
namespace WebCore {
using namespace VectorMath;
constint InputBufferSize = 8 * 16384;
// We only process the leading portion of the impulse response in the real-time thread. We don't exceed this length.
// It turns out then, that the background thread has about 278msec of scheduling slop.
// Empirically, this has been found to be a good compromise between giving enough time for scheduling slop,
// while still minimizing the amount of processing done in the primary (high-priority) thread.
// This was found to be a good value on Mac OS X, and may work well on other platforms as well, assuming
// the very rough scheduling latencies are similar on these time-scales. Of course, this code may need to be
// tuned for individual platforms if this assumption is found to be incorrect.
const size_t RealtimeFrameLimit = 8192 + 4096; // ~278msec @ 44.1KHz
const size_t MinFFTSize = 128;
const size_t MaxRealtimeFFTSize = 2048;
staticvoidbackgroundThreadEntry(void* threadData)
{
ReverbConvolver* reverbConvolver = static_cast<ReverbConvolver*>(threadData);
reverbConvolver->backgroundThreadEntry();
}
ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads)
: m_impulseResponseLength(impulseResponse->length())
, m_accumulationBuffer(impulseResponse->length() + renderSliceSize)
, m_inputBuffer(InputBufferSize)
, m_renderSliceSize(renderSliceSize)
, m_minFFTSize(MinFFTSize) // First stage will have this size - successive stages will double in size each time
, m_maxFFTSize(maxFFTSize) // until we hit m_maxFFTSize
, m_useBackgroundThreads(useBackgroundThreads)
, m_backgroundThread(0)
, m_wantsToExit(false)
, m_moreInputBuffered(false)
{
// If we are using background threads then don't exceed this FFT size for the
// stages which run in the real-time thread. This avoids having only one or two
// large stages (size 16384 or so) at the end which take a lot of time every several
// processing slices. This way we amortize the cost over more processing slices.
m_maxRealtimeFFTSize = MaxRealtimeFFTSize;
// For the moment, a good way to know if we have real-time constraint is to check if we're using background threads.
// Otherwise, assume we're being run from a command-line tool.
bool hasRealtimeConstraint = useBackgroundThreads;
constfloat* response = impulseResponse->data();
size_t totalResponseLength = impulseResponse->length();
// The total latency is zero because the direct-convolution is used in the leading portion.
size_t reverbTotalLatency = 0;
size_t stageOffset = 0;
int i = 0;
size_t fftSize = m_minFFTSize;
while (stageOffset < totalResponseLength) {
size_t stageSize = fftSize / 2;
// For the last stage, it's possible that stageOffset is such that we're straddling the end
// of the impulse response buffer (if we use stageSize), so reduce the last stage's length...
if (stageSize + stageOffset > totalResponseLength)
stageSize = totalResponseLength - stageOffset;
// This "staggers" the time when each FFT happens so they don't all happen at the same time
int renderPhase = convolverRenderPhase + i * renderSliceSize;
bool useDirectConvolver = !stageOffset;
OwnPtr<ReverbConvolverStage> stage = adoptPtr(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver));
bool isBackgroundStage = false;
if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
m_backgroundStages.append(stage.release());
isBackgroundStage = true;
} else
m_stages.append(stage.release());
stageOffset += stageSize;
++i;
if (!useDirectConvolver) {
// Figure out next FFT size
fftSize *= 2;
}
if (hasRealtimeConstraint && !isBackgroundStage && fftSize > m_maxRealtimeFFTSize)
fftSize = m_maxRealtimeFFTSize;
if (fftSize > m_maxFFTSize)
fftSize = m_maxFFTSize;
}
// Start up background thread
// FIXME: would be better to up the thread priority here. It doesn't need to be real-time, but higher than the default...
if (this->useBackgroundThreads() && m_backgroundStages.size() > 0)
m_backgroundThread = createThread(WebCore::backgroundThreadEntry, this, "convolution background thread");
}
ReverbConvolver::~ReverbConvolver()
{
// Wait for background thread to stop
if (useBackgroundThreads() && m_backgroundThread) {
m_wantsToExit = true;
// Wake up thread so it can return
{
MutexLocker locker(m_backgroundThreadLock);
m_moreInputBuffered = true;
m_backgroundThreadCondition.signal();
}
waitForThreadCompletion(m_backgroundThread);
}
}
voidReverbConvolver::backgroundThreadEntry()
{
while (!m_wantsToExit) {
// Wait for realtime thread to give us more input
m_moreInputBuffered = false;
{
MutexLocker locker(m_backgroundThreadLock);
while (!m_moreInputBuffered && !m_wantsToExit)
m_backgroundThreadCondition.wait(m_backgroundThreadLock);
}
// Process all of the stages until their read indices reach the input buffer's write index
int writeIndex = m_inputBuffer.writeIndex();
// Even though it doesn't seem like every stage needs to maintain its own version of readIndex
// we do this in case we want to run in more than one background thread.
int readIndex;
while ((readIndex = m_backgroundStages[0]->inputReadIndex()) != writeIndex) { // FIXME: do better to detect buffer overrun...
// The ReverbConvolverStages need to process in amounts which evenly divide half the FFT size
constint SliceSize = MinFFTSize / 2;
// Accumulate contributions from each stage
for (size_t i = 0; i < m_backgroundStages.size(); ++i)
m_backgroundStages[i]->processInBackground(this, SliceSize);
}
}
}
voidReverbConvolver::process(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess)
{
bool isSafe = sourceChannel && destinationChannel && sourceChannel->length() >= framesToProcess && destinationChannel->length() >= framesToProcess;
ASSERT(isSafe);
if (!isSafe)
return;
constfloat* source = sourceChannel->data();
float* destination = destinationChannel->mutableData();
bool isDataSafe = source && destination;
ASSERT(isDataSafe);
if (!isDataSafe)
return;
// Feed input buffer (read by all threads)
m_inputBuffer.write(source, framesToProcess);
// Accumulate contributions from each stage
for (size_t i = 0; i < m_stages.size(); ++i)
m_stages[i]->process(source, framesToProcess);
// Finally read from accumulation buffer
m_accumulationBuffer.readAndClear(destination, framesToProcess);
// Now that we've buffered more input, wake up our background thread.
// Not using a MutexLocker looks strange, but we use a tryLock() instead because this is run on the real-time
// thread where it is a disaster for the lock to be contended (causes audio glitching). It's OK if we fail to
// signal from time to time, since we'll get to it the next time we're called. We're called repeatedly
// and frequently (around every 3ms). The background thread is processing well into the future and has a considerable amount of
// leeway here...
if (m_backgroundThreadLock.tryLock()) {
m_moreInputBuffered = true;
m_backgroundThreadCondition.signal();
m_backgroundThreadLock.unlock();
}
}
voidReverbConvolver::reset()
{
for (size_t i = 0; i < m_stages.size(); ++i)
m_stages[i]->reset();
for (size_t i = 0; i < m_backgroundStages.size(); ++i)
m_backgroundStages[i]->reset();
m_accumulationBuffer.reset();
m_inputBuffer.reset();
}
size_t ReverbConvolver::latencyFrames() const
{
return 0;
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)